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Commit Graph

202 Commits

Author SHA1 Message Date
wm4
ddd068491c Replace remaining avcodec_close() calls
This API isn't deprecated (yet?), but it's still inferior and harder to
use than avcodec_free_context().

Leave the call only in 1 case in af_lavcac3enc.c, where we apparently
seriously close and reopen the encoder for whatever reason.
2017-07-16 12:51:48 +02:00
wm4
b016760a28 ad_spdif: minor cleanups
Use avcodec_free_context() unstead of random other calls. Actually it
was already used in the second case, but calling avcodec_close() is
redundant.

Don't crash if allocating a codec context fails.
2017-07-10 16:40:52 +02:00
wm4
d5702d3b95 ad_lavc, vd_lavc, sd_lavc: consistently use avcodec_free_context()
Instead of various ad-hoc ways to achieve the same thing. (The API was
added only later.)
2017-07-06 16:25:42 +02:00
wm4
6489b112ad dec_audio, ad_lavc: change license to LGPL
All relevant authors of the current code have agreed.

As always, there are the usual historical artifacts that could be
mentioned. For example, there used to be a large number of decoders
by various authors who were not asked, but whose code was all 100%
removed. (Mostly due to FFmpeg providing all codecs.)

One point of contention is that Nick Kurshev might have refactored the
old audio decoder code in 2001. Basically, there are hints that it might
have been done by him, such as Arpi's commit message stating that the
code was imported from MPlayerXP (Nick's fork), or all the files having
his name in the "maintainer" field. On the other hand, the murky history
of ad.h weakens this - it could be that Arpi started this work, and Nick
took it (and possibly finished it).

In any case, Nick could not be reached, so there is no agreement for
LGPL relicensing from him. We're changing the license anyway, and assume
that his change in itself is not copyrightable. He only moved code, and
in addition used the equivalent video decoder framework (done by Arpi,
who agreed) as template. For example, ad_functions_s was basically
vd_functions_s, which the signature of the decode callback changed to
the same as audio_decode(). ad_functions_s also had a comment that said
it interfaces with "video decoder drivers" (I'm fixing this comment in
this commit).

I verified that no additional code was added that is copyright-relevant,
still in today's code, and not copied from the existing code at the time
(either from the previous audio decoder code or the video framework
code). What apparently matters here is that none of the old code was not
written by Nick, and the authors of the old code have given his
agreement, and (probably) that Nick didn't add actual new code (none
that would have survived), that was not trivially based on the old one
(i.e. no new copyrightable "work").

A copyright expert told me that this kind of change can be considered
not relevant for copyright, so here we go.

Rewriting this would end with the same code anyway, and the naming
conventions can't be copyrighted.
2017-06-14 21:08:59 +02:00
wm4
e77ed53459 ad_spdif: change license to LGPL
All authors have agreed. (Even the main author, if you wonder about the
entry in the Copyright file.)
2017-05-21 12:35:53 +02:00
wm4
380bc03823 ad.h: change license to LGPL
All authors have agreed.

Commit 94d3170bd0 is a bit murky: Nick could not be reached, and arpi's
changes were obviously inspired or copied from Nick's. However, the
changed symbols were removed and do not exist anymore.
2017-05-05 07:32:35 +02:00
wm4
e50e9b6120 dec_video, dec_audio: remove redundant NULL-checks
OK, they're redundant. Now stop wasting my time, coverity.
2017-02-20 13:58:18 +01:00
wm4
96a45a16af player: add experimental stream recording feature
This is basically a WIP, but it can't remain in a branch forever. A
warning is print when using it as it's still a bit "shaky".
2017-02-07 17:05:17 +01:00
wm4
801fa486b0 ad_lavc, vd_lavc: move mpv->lavc decoder parameter setup to common code
This can be useful in other contexts.

Note that we end up setting AVCodecContext.width/height instead of
coded_width/coded_height now. AVCodecParameters can't set coded_width,
but this is probably more correct anyway.
2017-01-25 08:24:19 +01:00
wm4
b14fac9afa build: replace some FFmpeg API checks with version checks
The FFmpeg versions we support all have the APIs we were checking for.
Only Libav missed them. Simplify this by explicitly checking for FFmpeg
in the code, instead of trying to detect the presence of the API.
2017-01-24 08:11:42 +01:00
wm4
6be58df8d1 ad_lavc: respect AV_FRAME_FLAG_DISCARD
Since we set "skip_manual", we can actually get frames with this set.
Currently, only AV_PKT_FLAG_DISCARD will trigger this flag, and only
mov.c sets the latter flags, so this is related to FFmpeg's half-broken
mp4 edit list support.
2017-01-24 08:04:53 +01:00
wm4
8cbb2b5e9a ad_spdif: log avformat errors 2017-01-19 12:44:28 +01:00
wm4
c522d0dfbd ad_spdif: fix obscure cases of AC3 passthrough
Apparently you set the native sample rate when passing through AC3.
This fixes passthrough with 44100 Hz AC3.

Avoid opening a decoder for this and only open the parser. (Hopefully
DTS will also support this some time in the future or so - having to
open a decoder just to get the profile is dumb.)
2017-01-18 10:22:28 +01:00
wm4
cbd8abcbff audio: restructure decode loop
Same deal as with video. Including the EOF handling.

(It would be nice if this code were not duplicated, but right now we're
not even close to unifying the audio and video code paths.)
2017-01-11 11:58:32 +01:00
wm4
9d21f2503f options: deprecate codec family selection in --vd/--ad
Useless now, so get rid of it. Also affects some user-visible display
things (like reported codec in use).
2016-12-23 18:12:29 +01:00
wm4
c560f6ff0a audio: change how spdif codecs are selected
Remove ad_spdif from the normal codec list, and select it explicitly.

One goal was to decouple this from the normal codec selection, so
they're less entangled and the decoder selection code can be simplified
in the far future. This means spdif codec selection is now done
explicitly via select_spdif_codec(). We can also remove the weird
requirements on "dts" and "dts-hd" for the --audio-spdif option, and it
can just do the right thing.

Now both video and audio codecs consist of a single codec family each,
vd_lavc and ad_lavc.
2016-12-23 18:10:07 +01:00
wm4
e57037dc95 ad_lavc, vd_lavc: don't set AVCodecContext.refcounted_frames
This field is (or should be) deprecated, and there's no need to set it
with the new API.
2016-12-18 12:28:09 +01:00
Michael Forney
2d9b6ff7cd ad_spdif: Fix crash when spdif muxer is not available
Currently, if init_filter fails after lavf_ctx is allocated, uninit is called
which frees lavf_ctx, but doesn't clear the pointer in spdif_ctx. So, on the
next call of decode_packet, it thinks it is already initialized and uses it,
resulting in a crash on my system.
2016-12-11 14:20:58 +01:00
wm4
3eceac2eab Remove compatibility things
Possible with bumped FFmpeg/Libav.

These are just the simple cases.
2016-12-07 19:53:11 +01:00
wm4
67467103e8 dec_video, dec_audio: avoid full reinit on switches to the same segment
Same deal as with the previous commit.

(Unfortunately, this code is still duplicated.)
2016-11-09 16:44:06 +01:00
wm4
139f6b5de7 ad_lavc, vd_lavc: fix a recent libavcodec deprecation warning
Both AVFrame.pts and AVFrame.pkt_pts have existed for a long time. Until
now, decoders always returned the pts via the pkt_pts field, while the
pts field was used for encoding and libavfilter only. Recently, pkt_pts
was deprecated, and pts was switched to always carry the pts.

This means we have to be careful not to accidentally use the wrong
field, depending on the libavcodec version. We have to explicitly check
the version numbers. Of course the version numbers are completely
idiotic, because idiotically the pkg-config and library names are the
same for FFmpeg and Libav, so we have to deal with this explicitly as
well.
2016-10-17 19:18:03 +02:00
wm4
6f4d918cb7 audio: dump timestamp difference
Can help to analyze timestamp jitter or seeing completely bogus
timestamps.
2016-10-02 12:55:22 +02:00
wm4
0110b738d5 vd_lavc, ad_lavc: set pkt_timebase, not time_base
These are different AVCodecContext fields. pkt_timebase is the correct
one for identifying the unit of packet/frame timestamps when decoding,
while time_base is for encoding. Some decoders also overwrite the
time_base field with some unrelated codec metadata.

pkt_timebase does not exist in Libav, so an #if is required.
2016-08-29 12:46:12 +02:00
wm4
a47d849df7 ad_lavc: actually tell decoder about the timebase
Essentially forgotten in commit 05e4df3f.
2016-08-23 12:06:47 +02:00
wm4
05e4df3f0c video/audio: always provide "proper" timestamps to libavcodec
Instead of passing through double float timestamps opaquely, pass real
timestamps. Do so by always setting a valid timebase on the
AVCodecContext for audio and video decoding.

Specifically try not to round timestamps to a too coarse timebase, which
could round off small adjustments to timestamps (such as for start time
rebasing or demux_timeline). If the timebase is considered too coarse,
make it finer.

This gets rid of the need to do this specifically for some hardware
decoding wrapper. The old method of passing through double timestamps
was also a bit questionable. While libavcodec is not supposed to
interpret timestamps at all if no timebase is provided, it was
needlessly tricky. Also, it actually does compare them with
AV_NOPTS_VALUE. This change will probably also reduce confusion in the
future.
2016-08-19 14:59:30 +02:00
wm4
0b144eac39 audio: use --audio-channels=auto behavior, except on ALSA
This commit adds an --audio-channel=auto-safe mode, and makes it the
default. This mode behaves like "auto" with most AOs, except with
ao_alsa. The intention is to allow multichannel output by default on
sane APIs. ALSA is not sane as in it's so low level that it will e.g.
configure any layout over HDMI, even if the connected A/V receiver does
not support it. The HDMI fuckup is of course not ALSA's fault, but other
audio APIs normally isolate applications from dealing with this and
require the user to globally configure the correct output layout.

This will help with other AOs too. ao_lavc (encoding) is changed to the
new semantics as well, because it used to force stereo (perhaps because
encoding mode is supposed to produce safe files for crap devices?).
Exclusive mode output on Windows might need to be adjusted accordingly,
as it grants the same kind of low level access as ALSA (requires more
research).

In addition to the things mentioned above, the --audio-channels option
is extended to accept a set of channel layouts. This is supposed to be
the correct way to configure mpv ALSA multichannel output. You need to
put a list of channel layouts that your A/V receiver supports.
2016-08-04 20:49:20 +02:00
wm4
614efea3e6 ad_lavc: work around braindead ffmpeg behavior
The libavcodec wmapro decoder will skip some bytes at the start of the
first packet and return each time. It will not return any audio data in
this state.

Our own code as well as libavcodec's new API handling
(avcodec_send_packet() etc.) discard the PTS on the first return, which
means the PTS is never known for the first packet. This results in a
"Failed audio resync." message.

Fixy it by remember the PTS in next_pts. This field is used only if the
decoder outputs no PTS, and is updated after each frame - and thus
should be safe to set.

(Possibly this should be fixed in libavcodec new API handling by not
setting the PTS to NOPTS as long as no real data has been output. It
could even interpolate the PTS if the timebase is known.)

Fixes the failure message seen in #3297.
2016-07-01 15:51:34 +02:00
wm4
3e58ce96ac dec_audio: fix segment boudnary switching
Some bugs in this code are exposed by e.g. playing lossless audio files
with --ad-lavc-threads=16. (libavcodec doesn't really support threaded
audio decoding, except for lossless files.) In these cases, a major
amount of audio can be buffered, which makes incorrect handling of this
buffering obvious.

For one, draining the decoder can take a while, so if there's a new
segment, we shouldn't read audio.

The segment end check was completely wrong, and used the start value.
2016-06-27 15:12:21 +02:00
wm4
7ea22fe889 ad_lavc: resume from mid-stream EOF conditions with new decode API
Workaround for an awful corner-case. The new decode API "locks" the
decoder into the EOF state once a drain packet has been sent. The
problem starts with a file containing a 0-sized packet, which is
interpreted as drain packet.

This should probably be changed in libavcodec (not treating 0-sized
packets as drain packets with the new API) or in libavformat (discard
0-sized packets as invalid), but efforts to do so have been fruitless.

Note that vd_lavc.c already does something similar, but originally for
other reasons.

Fixes #3106.
2016-06-22 21:37:36 +02:00
wm4
78346e9c9a ad_spdif: take care of deprecated libavcodec API usage 2016-04-20 19:37:45 +02:00
wm4
c971220cdd demux_lavf, ad_lavc, ad_spdif, vd_lavc: handle FFmpeg codecpar API change
AVFormatContext.codec is deprecated now, and you're supposed to use
AVFormatContext.codecpar instead.

Handle this for all of the normal playback code.

Encoding mode isn't touched.
2016-03-31 22:00:45 +02:00
wm4
4300bfd518 ad_lavc, vd_lavc: support new Libav decoding API
For now only found in Libav.
2016-03-24 17:53:30 +01:00
wm4
f0febc35eb ad_lavc: add codec_timebase hack too
vd_lavc.c had this, and soon I'll need it in ad_lavc.c too. For now it's
unused.
2016-03-24 16:39:15 +01:00
wm4
7c181e5b9b audio: make mp_audio_skip_samples() adjust the PTS
Slight simplification/cleanup.
2016-02-22 20:13:31 +01:00
wm4
9ee340c3af ad_lavc: skip AVCodecContext.delay samples at beginning
Fixes correctness_trimming_nobeeps.opus. One nasty thing is that this
mechanism interferes with the container-signalled mechanism with
AV_FRAME_DATA_SKIP_SAMPLES. So apply it only if that is apparently not
present. It's a mess, and it's still broken in FFmpeg CLI, so I'm sure
this will get fucked up later again.
2016-02-22 20:10:38 +01:00
wm4
289edadb8d ad_lavc: make sample trimming symmetric to skipping
I'm not quite sure what the FFmpeg AV_FRAME_DATA_SKIP_SAMPLES API
demands here. The code so far assumed that skipping can be more than a
frame, but not trimming. Extend it to trimming too.
2016-02-22 19:58:11 +01:00
wm4
d52b2981c0 ad_lavc: move skipping logic out of the HAVE_AVFRAME_SKIP_SAMPLES block 2016-02-22 19:50:09 +01:00
wm4
65b858f7d3 ad_lavc: interpolate missing timestamps
This is actually already done by dec_audio.c. But if
AV_FRAME_DATA_SKIP_SAMPLES is applied, this happens too late here. The
problem is that this will slice off samples, and make it impossible for
later code to reconstruct the timestamp properly.

Missing timestamps can still happen with some demuxers, e.g. demux_mkv.c
with Opus tracks. (Although libavformat interpolates these itself.)
2016-02-22 13:08:36 +01:00
wm4
1bb1543a88 audio: move frame clipping to a generic function 2016-02-21 18:16:41 +01:00
wm4
0af5335383 Rewrite ordered chapters and timeline stuff
This uses a different method to piece segments together. The old
approach basically changes to a new file (with a new start offset) any
time a segment ends. This meant waiting for audio/video end on segment
end, and then changing to the new segment all at once. It had a very
weird impact on the playback core, and some things (like truly gapless
segment transitions, or frame backstepping) just didn't work.

The new approach adds the demux_timeline pseudo-demuxer, which presents
an uniform packet stream from the many segments. This is pretty similar
to how ordered chapters are implemented everywhere else. It also reminds
of the FFmpeg concat pseudo-demuxer.

The "pure" version of this approach doesn't work though. Segments can
actually have different codec configurations (different extradata), and
subtitles are most likely broken too. (Subtitles have multiple corner
cases which break the pure stream-concatenation approach completely.)

To counter this, we do two things:
- Reinit the decoder with each segment. We go as far as allowing
  concatenating files with completely different codecs for the sake
  of EDL (which also uses the timeline infrastructure). A "lighter"
  approach would try to make use of decoder mechanism to update e.g.
  the extradata, but that seems fragile.
- Clip decoded data to segment boundaries. This is equivalent to
  normal playback core mechanisms like hr-seek, but now the playback
  core doesn't need to care about these things.

These two mechanisms are equivalent to what happened in the old
implementation, except they don't happen in the playback core anymore.
In other words, the playback core is completely relieved from timeline
implementation details. (Which honestly is exactly what I'm trying to
do here. I don't think ordered chapter behavior deserves improvement,
even if it's bad - but I want to get it out from the playback core.)

There is code duplication between audio and video decoder common code.
This is awful and could be shareable - but this will happen later.

Note that the audio path has some code to clip audio frames for the
purpose of codec preroll/gapless handling, but it's not shared as
sharing it would cause more pain than it would help.
2016-02-15 21:04:07 +01:00
wm4
f2b039da77 audio/video: expose codec info as separate field
Preparation for the timeline rewrite. The codec will be able to change,
the stream header not.
2016-02-15 20:34:45 +01:00
wm4
6eae6a785c ad_lavc: fix --ad-lavc-threads range
The code is shared with the --vd-lavc-threads option, so using 0 for
auto-detection just works.

But no, this is not useful. Just change it for orthogonality.
2016-02-11 22:06:58 +01:00
wm4
bb6ae0e50b audio: minor simplification
These fields are already deallocated by uninit_decoder(). Also remove
the wrong/useless log message.
2016-02-05 23:43:25 +01:00
wm4
ab318aeea8 audio/video: merge decoder return values
Will be helpful for the coming filter support. I planned on merging
audio/video decoding, but this will have to wait a bit longer, so only
remove the duplicate status codes.
2016-02-01 22:03:04 +01:00
wm4
c5a48c6332 audio: move pts reset check
Reduces the dependency of the filter/output code on the decoder.
2016-01-29 22:44:20 +01:00
wm4
fef8b7984b audio: refactor: work towards unentangling audio decoding and filtering
Similar to the video path. dec_audio.c now handles decoding only. It
also looks very similar to dec_video.c, and actually contains some of
the rewritten code from it. (A further goal might be unifying the
decoders, I guess.)

High potential for regressions.
2016-01-22 00:25:44 +01:00
wm4
ca00e347fc ad_spdif: if DTS-HD is requested, and profile unknown, use DTS-HD
This means there will be no loss if profile detection failed for some
reason.
2016-01-20 17:18:28 +01:00
wm4
aaafbfcc06 audio: remove initial decoding retry limitation
Seems useless.

This only helped in one case: one audio stream in the sample
av_find_best_stream_fails.ts had a AC3 packets which couldn't be
decoded, and for which avcodec_decode_audio4() returned 0 forever. In
this specific case, playback will now not start, and you have to
deselect audio manually.

(If someone complains, the old behavior might be restored, but
differently.)

Also remove the stale "bitrate" field.
2016-01-19 22:49:05 +01:00
wm4
30031edce3 audio: move direct packet reading from decoders to common code
Another bit of preparation.
2016-01-19 22:24:38 +01:00
wm4
c365b44e19 audio: move dec_audio.pool to ad_spdif
That's where its only use is.
2016-01-19 21:33:05 +01:00